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Supported Languages

Supported languages, regional dialects, and local speech variants for Universal-3 Pro Streaming.

For the complete documentation index, see llms.txt

Supported languages

Universal-3 Pro Streaming supports 6 languages with native multilingual code switching. The model automatically detects and switches between languages mid-stream.

If you need support beyond the 6 languages listed here, consider using the Whisper Streaming model (speech_model: "whisper-rt"), which supports 99 languages with automatic language detection. See the Whisper Streaming page for details.

Regional dialects and variants

Universal-3 Pro Streaming goes beyond standard language support with deep understanding of regional dialects and local variants. Whether your audio features Quebecois French, Mexican Spanish, or Brazilian Portuguese, the model accurately captures speech as it's naturally spoken — including colloquial expressions, local vocabulary, and accent-specific pronunciation patterns.

You do not need to specify a dialect code to get accurate dialect transcription. Universal-3 Pro automatically recognizes regional speech patterns when using the base language code (e.g., fr for all French dialects, es for all Spanish dialects).

Dialect / Variant	Description
American English	Standard US English, including regional variants (Southern, Midwestern, Northeastern)
British English	UK English, including Received Pronunciation and regional accents
Australian English	Australian English with local expressions and pronunciation

Dialect / Variant	Description
Castilian Spanish	Standard Peninsular Spanish as spoken in central and northern Spain
Mexican Spanish	Mexican Spanish with local vocabulary and pronunciation
Argentine Spanish	Rioplatense Spanish with distinctive voseo and pronunciation
Colombian Spanish	Colombian Spanish with regional speech patterns
Chilean Spanish	Chilean Spanish with rapid speech patterns and local slang
Caribbean Spanish	Cuban, Dominican, and Puerto Rican Spanish dialects
Spanglish	English-Spanish code-mixing common in US bilingual communities

Dialect / Variant	Description
Metropolitan French	Standard Parisian French
Canadian French (Quebecois)	Quebec French with distinctive vocabulary, pronunciation, and expressions
Belgian French	Belgian French with local vocabulary and pronunciation

Dialect / Variant	Description
Brazilian Portuguese	Brazilian Portuguese with local vocabulary, pronunciation, and expressions
European Portuguese	Standard Lisbon Portuguese with Iberian pronunciation

Dialect / Variant	Description
Standard Italian	Standard Italian based on Tuscan-influenced speech

Configuration

Prompting can be used to guide the model toward a specific language. However, prompts should be thoroughly tested before use in production. For example, pre-pending Transcribe Spanish to your prompt has shown to perform well on Spanish audio. See the Prompting guide for more details.

Quickstart

The following examples demonstrate how to stream audio to Universal-3 Pro Streaming.

Install the required libraries

pip install "assemblyai>=0.54.0" pyaudio

Create a new file main.py and paste the code below. Replace `` with your API key.

Run with python main.py and speak into your microphone.

import logging
from typing import Type

import assemblyai as aai
from assemblyai.streaming.v3 import (
    BeginEvent,
    StreamingClient,
    StreamingClientOptions,
    StreamingError,
    StreamingEvents,
    StreamingParameters,
    TurnEvent,
    TerminationEvent,
)

api_key = "<YOUR_API_KEY>"

logging.basicConfig(level=logging.INFO)
logger = logging.getLogger(__name__)

def on_begin(self: Type[StreamingClient], event: BeginEvent):
    print(f"Session started: {event.id}")

def on_turn(self: Type[StreamingClient], event: TurnEvent):
    print(f"{event.transcript} ({event.end_of_turn})")

def on_terminated(self: Type[StreamingClient], event: TerminationEvent):
    print(
        f"Session terminated: {event.audio_duration_seconds} seconds of audio processed"
    )

def on_error(self: Type[StreamingClient], error: StreamingError):
    print(f"Error occurred: {error}")

def main():
    client = StreamingClient(
        StreamingClientOptions(
            api_key=api_key,
            api_host="streaming.assemblyai.com",
        )
    )

    client.on(StreamingEvents.Begin, on_begin)
    client.on(StreamingEvents.Turn, on_turn)
    client.on(StreamingEvents.Termination, on_terminated)
    client.on(StreamingEvents.Error, on_error)

    client.connect(
        StreamingParameters(
            sample_rate=16000,
            speech_model="u3-rt-pro",
        )
    )

    try:
        client.stream(
            aai.extras.MicrophoneStream(sample_rate=16000)
        )
    finally:
        client.disconnect(terminate=True)

if __name__ == "__main__":
    main()

Install the required libraries

pip install websocket-client pyaudio

Create a new file main.py and paste the code below. Replace `` with your API key.

Run with python main.py and speak into your microphone.

import pyaudio
import websocket
import json
import threading
import time
from urllib.parse import urlencode

YOUR_API_KEY = "<YOUR_API_KEY>"

CONNECTION_PARAMS = {
    "sample_rate": 16000,
    "speech_model": "u3-rt-pro",
}
API_ENDPOINT_BASE_URL = "wss://streaming.assemblyai.com/v3/ws"
API_ENDPOINT = f"{API_ENDPOINT_BASE_URL}?{urlencode(CONNECTION_PARAMS)}"

FRAMES_PER_BUFFER = 800
SAMPLE_RATE = CONNECTION_PARAMS["sample_rate"]
CHANNELS = 1
FORMAT = pyaudio.paInt16

audio = None
stream = None
ws_app = None
audio_thread = None
stop_event = threading.Event()

def on_open(ws):
    print("WebSocket connection opened.")
    def stream_audio():
        global stream
        while not stop_event.is_set():
            try:
                audio_data = stream.read(FRAMES_PER_BUFFER, exception_on_overflow=False)
                ws.send(audio_data, websocket.ABNF.OPCODE_BINARY)
            except Exception as e:
                print(f"Error streaming audio: {e}")
                break

    global audio_thread
    audio_thread = threading.Thread(target=stream_audio)
    audio_thread.daemon = True
    audio_thread.start()

def on_message(ws, message):
    try:
        data = json.loads(message)
        msg_type = data.get("type")

        if msg_type == "Begin":
            print(f"Session began: ID={data.get('id')}")
        elif msg_type == "Turn":
            transcript = data.get("transcript", "")
            end_of_turn = data.get("end_of_turn", False)
            if end_of_turn:
                print(f"\r{' ' * 80}\r{transcript}")
            else:
                print(f"\r{transcript}", end="")
        elif msg_type == "Termination":
            print(f"\nSession terminated: {data.get('audio_duration_seconds', 0)}s of audio")
    except Exception as e:
        print(f"Error handling message: {e}")

def on_error(ws, error):
    print(f"\nWebSocket Error: {error}")
    stop_event.set()

def on_close(ws, close_status_code, close_msg):
    print(f"\nWebSocket Disconnected: Status={close_status_code}")
    global stream, audio
    stop_event.set()
    if stream:
        if stream.is_active():
            stream.stop_stream()
        stream.close()
    if audio:
        audio.terminate()

def run():
    global audio, stream, ws_app

    audio = pyaudio.PyAudio()
    stream = audio.open(
        input=True,
        frames_per_buffer=FRAMES_PER_BUFFER,
        channels=CHANNELS,
        format=FORMAT,
        rate=SAMPLE_RATE,
    )
    print("Speak into your microphone. Press Ctrl+C to stop.")

    ws_app = websocket.WebSocketApp(
        API_ENDPOINT,
        header={"Authorization": YOUR_API_KEY},
        on_open=on_open,
        on_message=on_message,
        on_error=on_error,
        on_close=on_close,
    )

    ws_thread = threading.Thread(target=ws_app.run_forever)
    ws_thread.daemon = True
    ws_thread.start()

    try:
        while ws_thread.is_alive():
            time.sleep(0.1)
    except KeyboardInterrupt:
        print("\nStopping...")
        stop_event.set()
        if ws_app and ws_app.sock and ws_app.sock.connected:
            ws_app.send(json.dumps({"type": "Terminate"}))
            time.sleep(2)
        if ws_app:
            ws_app.close()
        ws_thread.join(timeout=2.0)

if __name__ == "__main__":
    run()

Install the required libraries

npm install assemblyai node-record-lpcm16

The module node-record-lpcm16 requires SoX and it must be available in your $PATH.

For Mac OS:

brew install sox

For most linux distros:

sudo apt-get install sox libsox-fmt-all

For Windows:

download the binaries

Create a new file main.js and paste the code below. Replace `` with your API key.

Run with node main.js and speak into your microphone.

import { Readable } from "stream";
import { AssemblyAI } from "assemblyai";
import recorder from "node-record-lpcm16";

const run = async () => {
  const client = new AssemblyAI({
    apiKey: "<YOUR_API_KEY>",
  });

  const transcriber = client.streaming.transcriber({
    sampleRate: 16_000,
    speechModel: "u3-rt-pro",
  });

  transcriber.on("open", ({ id }) => {
    console.log(`Session opened with ID: ${id}`);
  });

  transcriber.on("error", (error) => {
    console.error("Error:", error);
  });

  transcriber.on("close", (code, reason) =>
    console.log("Session closed:", code, reason)
  );

  transcriber.on("turn", (turn) => {
    if (!turn.transcript) {
      return;
    }

    console.log("Turn:", turn.transcript);
  });

  try {
    console.log("Connecting to streaming transcript service");
    await transcriber.connect();

    console.log("Starting recording");
    const recording = recorder.record({
      channels: 1,
      sampleRate: 16_000,
      audioType: "wav",
    });

    Readable.toWeb(recording.stream()).pipeTo(transcriber.stream());

    process.on("SIGINT", async function () {
      console.log();
      console.log("Stopping recording");
      recording.stop();

      console.log("Closing streaming transcript connection");
      await transcriber.close();

      process.exit();
    });
  } catch (error) {
    console.error(error);
  }
};

run();

Install the required libraries

npm install ws mic

Create a new file main.js and paste the code below. Replace `` with your API key.

Run with node main.js and speak into your microphone.

const WebSocket = require("ws");
const mic = require("mic");
const querystring = require("querystring");

const YOUR_API_KEY = "<YOUR_API_KEY>";
const CONNECTION_PARAMS = {
  sample_rate: 16000,
  speech_model: "u3-rt-pro",
};
const API_ENDPOINT_BASE_URL = "wss://streaming.assemblyai.com/v3/ws";
const API_ENDPOINT = `${API_ENDPOINT_BASE_URL}?${querystring.stringify(CONNECTION_PARAMS)}`;

const SAMPLE_RATE = CONNECTION_PARAMS.sample_rate;

let micInstance = null;
let ws = null;

function run() {
  console.log("Starting AssemblyAI streaming transcription...");

  ws = new WebSocket(API_ENDPOINT, {
    headers: { Authorization: YOUR_API_KEY },
  });

  ws.on("open", () => {
    console.log("WebSocket connection opened.");

    micInstance = mic({
      rate: String(SAMPLE_RATE),
      channels: "1",
      bitwidth: "16",
      encoding: "signed-integer",
      endian: "little",
    });

    const micInputStream = micInstance.getAudioStream();
    micInputStream.on("data", (data) => {
      if (ws.readyState === WebSocket.OPEN) {
        ws.send(data);
      }
    });

    micInstance.start();
    console.log("Speak into your microphone. Press Ctrl+C to stop.");
  });

  ws.on("message", (data) => {
    try {
      const msg = JSON.parse(data);
      if (msg.type === "Begin") {
        console.log(`Session began: ID=${msg.id}`);
      } else if (msg.type === "Turn") {
        const transcript = msg.transcript || "";
        if (msg.end_of_turn) {
          process.stdout.write("\r" + " ".repeat(80) + "\r");
          console.log(transcript);
        } else {
          process.stdout.write(`\r${transcript}`);
        }
      } else if (msg.type === "Termination") {
        console.log(
          `\nSession terminated: ${msg.audio_duration_seconds}s of audio`
        );
      }
    } catch (e) {
      console.error("Error parsing message:", e);
    }
  });

  ws.on("error", (error) => {
    console.error("WebSocket error:", error);
  });

  ws.on("close", (code, reason) => {
    console.log(`WebSocket closed: ${code}`);
    if (micInstance) micInstance.stop();
  });

  process.on("SIGINT", () => {
    console.log("\nStopping...");
    if (micInstance) micInstance.stop();
    if (ws && ws.readyState === WebSocket.OPEN) {
      ws.send(JSON.stringify({ type: "Terminate" }));
      setTimeout(() => ws.close(), 2000);
    }
  });
}

run();